1. Field of the Invention
The present invention relates to a network, a private branch exchange, and a PBX additional service starting method, particularly to a terminal connected to a network including an Internet protocol private branch exchange (IP-PBX).
2. Related Background Art
This type of network has heretofore been constituted by connection of a private branch exchange to a fixed phone terminal, a radio terminal such as a personal handy-phone system (PHS), a voice over Internet protocol (VoIP) terminal or the like via a local area network (LAN).
As the VoIP terminal, in addition to an IP phone terminal and an Internet phone terminal, there is a session initiation protocol terminal. The SIP is a communication protocol for use in starting or ending multimedia communication such as sound communication (fixed phone, cellular phone, etc.), video communication such as television phone, chat (conversation by characters) and the like in an environment of an IP network using data having a form referred to as an IP packet.
In the IP network, in general, connection-less type communication is performed without confirming connection to a target as in electronic mails. On the other hand, in the fixed phone, in general, connection type communication is performed while confirming the connection with the target. The SIP realizes the connection type communication in the IP network.
The SIP basically comprises methods (operations) such as INVITE (session between users is established), ACK (acknowledgment), CANCEL (INVITE is ended during the establishment of the session), and BYE (the end of the session). The respective methods are exchanged as requests and responses to the requests between clients and servers to thereby establish or end the session.
Moreover, the SIP has characteristics that applications can be comparatively easily prepared. For example, when a new service is added to H.323 of ITU-T for use in the IP phone, an H.450.x protocol which defines the H.323 additional service is added, and all H.323 end points on the network and software of a gate keeper need to be updated. However, in the SIP, an SIP application server which provides the new service is added, and the corresponding application is added. Then, the new service is usable.
In a conventional SIP terminal, when a call is transferred during calling, two types of transfer services can be provided [simple operation (unattended transfer) (a transfer method in which a transferor resets before a response of a transfer target)/completely automatic transfer (attended transfer) (a transfer method in which the transferor resets after the transfer target makes the response)]. For example, when the simple operation is used as the transfer service in the conventional SIP terminal, as shown in FIGS. 12 to 14, the call is transferred to SIP corresponding terminal #3 during the calling between SIP corresponding terminals #1 and #2.
When a key operation of the simple operation is performed (f2 of FIG. 12) during the calling between the SIP corresponding terminals #1 and #2 (f1 of FIG. 12), the SIP corresponding terminal #2 sends “REFER” to the SIP corresponding terminal #1 (f3 of FIG. 12). In response to this, the SIP corresponding terminal #1 returns “202 Accepted” to the SIP corresponding terminal #2 (f4 of FIG. 12), and sends “NOTIFY” (f5 of FIG. 12), and therefore the SIP corresponding terminal #2 returns “200 OK” to the SIP corresponding terminal #1 (f6 of FIG. 12).
Thereafter, the SIP corresponding terminal #2 disconnects the calling with the SIP corresponding terminal #1 (f7 of FIG. 12), and sends “BYE” to the SIP corresponding terminal #1 (f8 of FIG. 12). The SIP corresponding terminal #1 sends “200 OK” to the SIP corresponding terminal #2 (f9 of FIG. 12), and thereafter sends “INVITE (w/SDP)” to the SIP corresponding terminal #3 (f10 of FIG. 12).
After sending “180 Ringing” to the SIP corresponding terminal #1 (f11 of FIG. 12), the SIP corresponding terminal #3 responds to the calling (f12 of FIG. 12), and sends “200 OK (w/SDP)” to the SIP corresponding terminal #1 (f13 of FIG. 12). In response to this, the SIP corresponding terminal #1 returns “ACK” to the SIP corresponding terminal #3 (f14 of FIG. 12), and then the SIP corresponding terminals #1 and #3 enters the calling (f15 of FIG. 12).
Thereafter, since the SIP corresponding terminal #1 sends “NOTIFY” to the SIP corresponding terminal #2 (f16 of FIG. 12), the SIP corresponding terminal #2 returns “200 OK” to the SIP corresponding terminal #1 (f17 of FIG. 12).
On the other hand, when a key operation of completely automatic transfer is performed (g2 of FIG. 13) during the calling between the SIP corresponding terminals #1 and #2 (g1 of FIG. 13), the SIP corresponding terminal #2 sends “INVITE (Hold)” to the SIP corresponding terminal #1 (g3 of FIG. 13). When the “200 OK” returns from the SIP corresponding terminal #1 (g4 of FIG. 13), the SIP corresponding terminal #2 returns “ACK” to the SIP corresponding terminal #1 (g5 of FIG. 13), and sends “INVITE (w/SDP)” to the SIP corresponding terminal #3 (g6 of FIG. 13).
The SIP corresponding terminal #3 returns “180 Ringing” to the SIP corresponding terminal #2 (g7 of FIG. 13), responds to the calling with the SIP corresponding terminal #2 (g8 of FIG. 13), and returns “200 OK (w/SDP)” to the SIP corresponding terminal #2 (g9 of FIG. 13). When the SIP corresponding terminal #2 returns “ACK” to the SIP corresponding terminal #3 (g10 of FIG. 13), the SIP corresponding terminals #2 and #3 enter the calling (g11 of FIG. 13).
Thereafter, when disconnected (g12 of FIG. 13), the SIP corresponding terminal #2 sends “INVITE (Hold)” to the SIP corresponding terminal #3 (g13 of FIG. 13). Then, the SIP corresponding terminal #3 returns “200 OK (w/SDP)” to the SIP corresponding terminal #2 (g14 of FIG. 13). Therefore, the SIP corresponding terminal #2 returns “ACK” to the SIP corresponding terminal #3 (g15 of FIG. 13), and sends “REFER” to the SIP corresponding terminal #1 (g16 of FIG. 13).
The SIP corresponding terminal #1 returns “202 Accepted” to the SIP corresponding terminal #2 (g17 of FIG. 13), and also sends “NOTIFY” (g18 of FIG. 13). In response to this, the SIP corresponding terminal #1 sends “INVITE (w/SDP)” to the SIP corresponding terminal #3 (g20 of FIG. 14).
Since the SIP corresponding terminal #3 returns “200 OK (w/SDP)” to the SIP corresponding terminal #1 (g21 of FIG. 14), the SIP corresponding terminal #1 sends “ACK” to the SIP corresponding terminal #3 (g22 of FIG. 14), and the SIP corresponding terminals #1 and #3 enter the calling (g23 of FIG. 14).
When the SIP corresponding terminal #3 sends “BYE” to the SIP corresponding terminal #2 (g24 of FIG. 14), the SIP corresponding terminal #2 returns “200 OK” to the SIP corresponding terminal #3 (g25 of FIG. 14), the SIP corresponding terminal #1 sends “NOTIFY” to the SIP corresponding terminal #2 (g26 of FIG. 14), and then the SIP corresponding terminal #2 returns “200 OK” to the SIP corresponding terminal #1 (g27 of FIG. 14).
When these processes end, the SIP corresponding terminal #2 sends “BYE” to the SIP corresponding terminal #1 (g28 of FIG. 14), and the SIP corresponding terminal #1 returns “200 OK” to the SIP corresponding terminal #2 (g29 of FIG. 14).
Here, “INVITE” indicates a method for use in establishing the session among participants, “180 RINGING” indicates the ringing, “200 OK” indicates that the response is successful, “ACK” indicates a method for use in permitting the establishment of the session, “REFER” indicates a method indicating reference, “Accepted” indicates a method for indicating acceptance, “NOTIFY” is a method for returning the present state information, and “BYE” indicates a method for ending the session.
Japanese Patent Application Laid-Open No. 11-331371, Japanese Patent Publication No. 2003-502945, Japanese Patent Publication No. 2003-517764, and “SIP: Session Initiation Protocol” [RFC (Request for Comments) 3261, June 2002, 8th to 34th pages] disclose the earlier techniques as described above.
However, in the communication between the SIP terminals, there is a problem that only services (e.g., transfer service, service for calling among three, call waiting service, etc.) defined by Internet engineering task force (IETF) can be provided.
That is, even when the SIP terminals are connected to the above-described network in the communication between the SIP terminals, there is a problem that various services (e.g., callback service, extension interruption service, third party control service, etc.) provided by the private branch exchange cannot be received.
Moreover, there is a problem that version of software of the SIP terminal (the above-described client) itself needs to be upgraded to thereby add the service in a case where the new service is added to the communication between the SIP terminals.
When the call is transferred in the conventional SIP terminal during the calling, as described above, two types of transfer services (simple operation/completely automatic transfer) can be provided, but these transfer services have different sequences, and therefore the SIP terminal itself needs to be conscious of either transfer service to be used.